[asterisk-dev] [Code Review] 2864: Testsuite: Fix sip_attended_transfer test for Asterisk 12+

[svnbot]

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(Updated Sept. 26, 2013, 12:31 p.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
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Committed in revision 4219


Bugs: ASTERISK-22531
    https://issues.asterisk.org/jira/browse/ASTERISK-22531


Repository: testsuite


Description
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sip_attended_transfer test was failing in Asterisk 12 because it was listening for nonexistent AMI events. Now it uses BridgeEnter and AttendedTransfer events in order to determine if the test was successful.

PEP-8: The new code should be PEP-8 compliant. I did not change pre-existing code though. The only change to pre-existing code I made was to add docstrings to some event callbacks since it was not immediately clear what they were doing.


Diffs
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  /asterisk/trunk/tests/channels/SIP/sip_attended_transfer/configs/ast1/sip.conf 4126 
  /asterisk/trunk/tests/channels/SIP/sip_attended_transfer/run-test 4126 

Diff: https://reviewboard.asterisk.org/r/2864/diff/


Testing
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Ran test multiple times against both Asterisk 11 and Asterisk 12. Test passes for me.


Thanks,

Mark Michelson

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